gst/rtp/: Added G726 pay/depayloaders. Fixes #538891.

Original commit message from CVS:
Patch by: mersad <mersad at axis dot com>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_base_init),
(gst_rtp_g726_depay_class_init), (gst_rtp_g726_depay_init),
(gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process),
(gst_rtp_g726_depay_plugin_init):
* gst/rtp/gstrtpg726depay.h:
* gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_base_init),
(gst_rtp_g726_pay_class_init), (gst_rtp_g726_pay_init),
(gst_rtp_g726_pay_setcaps), (gst_rtp_g726_pay_plugin_init):
* gst/rtp/gstrtpg726pay.h:
Added G726 pay/depayloaders. Fixes #538891.
This commit is contained in:
mersad 2008-06-18 10:12:57 +00:00 committed by Wim Taymans
parent 198224ef58
commit e3141bbb49
7 changed files with 517 additions and 0 deletions

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@ -1,3 +1,20 @@
2008-06-18 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: mersad <mersad at axis dot com>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_base_init),
(gst_rtp_g726_depay_class_init), (gst_rtp_g726_depay_init),
(gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process),
(gst_rtp_g726_depay_plugin_init):
* gst/rtp/gstrtpg726depay.h:
* gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_base_init),
(gst_rtp_g726_pay_class_init), (gst_rtp_g726_pay_init),
(gst_rtp_g726_pay_setcaps), (gst_rtp_g726_pay_plugin_init):
* gst/rtp/gstrtpg726pay.h:
Added G726 pay/depayloaders. Fixes #538891.
2008-06-17 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp/URLS:

View file

@ -17,6 +17,8 @@ libgstrtp_la_SOURCES = \
gstrtppcmudepay.c \
gstrtppcmupay.c \
gstrtppcmapay.c \
gstrtpg726pay.c \
gstrtpg726depay.c \
gstrtpg729pay.c \
gstrtpg729depay.c \
gstrtpgsmdepay.c \
@ -75,6 +77,8 @@ noinst_HEADERS = \
gstrtppcmudepay.h \
gstrtppcmupay.h \
gstrtppcmapay.h \
gstrtpg726depay.h \
gstrtpg726pay.h \
gstrtpg729depay.h \
gstrtpg729pay.h \
gstrtpgsmdepay.h \

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@ -31,6 +31,8 @@
#include "gstrtppcmapay.h"
#include "gstrtppcmadepay.h"
#include "gstrtppcmudepay.h"
#include "gstrtpg726depay.h"
#include "gstrtpg726pay.h"
#include "gstrtpg729depay.h"
#include "gstrtpg729pay.h"
#include "gstrtpgsmpay.h"
@ -86,6 +88,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_ilbc_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g726_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g726_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g729_depay_plugin_init (plugin))
return FALSE;

207
gst/rtp/gstrtpg726depay.c Normal file
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@ -0,0 +1,207 @@
/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
* Copyright (C) 2005 Zeeshan Ali <zeenix@gmail.com>
* Copyright (C) 2008 Axis Communications <dev-gstreamer@axis.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg726depay.h"
#define DEFAULT_BIT_RATE 32000
#define SAMPLE_RATE 8000
#define LAYOUT_G726 "g726"
/* elementfactory information */
static const GstElementDetails gst_rtp_g726depay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
"Extracts G.726 audio from RTP packets",
"Axis Communications <dev-gstreamer@axis.com>");
/* RtpG726Depay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate gst_rtp_g726_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"encoding-name = (string) { \"G726\", \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\"}, "
"clock-rate = (int) 8000;")
);
static GstStaticPadTemplate gst_rtp_g726_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
"channels = (int) 1, "
"rate = (int) 8000, "
"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
"layout = (string) \"g726\"")
);
static GstBuffer *gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpG726Depay, gst_rtp_g726_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtp_g726_depay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g726_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g726_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_g726depay_details);
}
static void
gst_rtp_g726_depay_class_init (GstRtpG726DepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertpdepayload_class->process = gst_rtp_g726_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_g726_depay_setcaps;
}
static void
gst_rtp_g726_depay_init (GstRtpG726Depay * rtpG726depay,
GstRtpG726DepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtpG726depay);
gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
gint clock_rate = 8000; /* default */
const gchar *encoding_name;
gint bitrate;
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "clock-rate", &clock_rate);
depayload->clock_rate = clock_rate;
encoding_name = gst_structure_get_string (structure, "encoding-name");
if (encoding_name == NULL || g_ascii_strcasecmp (encoding_name, "G726") == 0) {
bitrate = DEFAULT_BIT_RATE;
} else if (g_ascii_strcasecmp (encoding_name, "G726-16") == 0) {
bitrate = 16000;
} else if (g_ascii_strcasecmp (encoding_name, "G726-24") == 0) {
bitrate = 24000;
} else if (g_ascii_strcasecmp (encoding_name, "G726-32") == 0) {
bitrate = 32000;
} else if (g_ascii_strcasecmp (encoding_name, "G726-40") == 0) {
bitrate = 40000;
} else {
GST_WARNING ("Could not determine bitrate from encoding-name (%s)",
encoding_name);
ret = FALSE;
goto done;
}
GST_DEBUG ("RTP G.726 depayloader, bitrate set to %d\n", bitrate);
srccaps = gst_caps_new_simple ("audio/x-adpcm",
"channels", G_TYPE_INT, 1,
"rate", G_TYPE_INT, clock_rate,
"bitrate", G_TYPE_INT, bitrate,
"layout", G_TYPE_STRING, LAYOUT_G726, NULL);
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
done:
return ret;
}
static GstBuffer *
gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstCaps *srccaps;
GstBuffer *outbuf = NULL;
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf),
gst_rtp_buffer_get_marker (buf),
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
if (!srccaps) {
/* Set the default caps */
srccaps = gst_caps_new_simple ("audio/x-adpcm",
"channels", G_TYPE_INT, 1,
"rate", G_TYPE_INT, SAMPLE_RATE,
"bitrate", G_TYPE_INT, DEFAULT_BIT_RATE,
"layout", G_TYPE_STRING, LAYOUT_G726, NULL);
gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
}
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
return outbuf;
}
gboolean
gst_rtp_g726_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg726depay",
GST_RANK_MARGINAL, GST_TYPE_RTP_G726_DEPAY);
}

51
gst/rtp/gstrtpg726depay.h Normal file
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@ -0,0 +1,51 @@
/* GStreamer
* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
* Copyright (C) 2008 Axis Communications AB <dev-gstreamer@axis.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTP_G726_DEPAY_H__
#define __GST_RTP_G726_DEPAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
typedef struct _GstRtpG726Depay GstRtpG726Depay;
typedef struct _GstRtpG726DepayClass GstRtpG726DepayClass;
#define GST_TYPE_RTP_G726_DEPAY \
(gst_rtp_g726_depay_get_type())
#define GST_RTP_G726_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G726_DEPAY,GstRtpG726Depay))
#define GST_RTP_G726_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G726_DEPAY,GstRtpG726DepayClass))
#define GST_IS_RTP_G726_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G726_DEPAY))
#define GST_IS_RTP_G726_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G726_DEPAY))
struct _GstRtpG726Depay
{
GstBaseRTPDepayload depayload;
};
struct _GstRtpG726DepayClass
{
GstBaseRTPDepayloadClass parent_class;
};
gboolean gst_rtp_g726_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_G726_DEPAY_H__ */

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/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
* Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com>
* Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg726pay.h"
static const GstElementDetails gst_rtp_g726_pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload-encodes G.726 audio into a RTP packet",
"Axis Communications <dev-gstreamer@axis.com>");
static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
"channels = (int) 1, "
"rate = (int) 8000, "
"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
"layout = (string) \"g726\"; "
"audio/G723, channels=(int)1, rate=(int)8000; "
"audio/32KADPCM, channels=(int)1, rate=(int)8000")
);
static GstStaticPadTemplate gst_rtp_g726_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\" } ")
);
static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g726_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_g726_pay_details);
}
static void
gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
}
static void
gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay);
GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
/* sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
}
static gboolean
gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gchar *encoding_name;
GstStructure *structure = gst_caps_get_structure (caps, 0);
const gchar *stname = gst_structure_get_name (structure);
GstBaseRTPAudioPayload *basertpaudiopayload;
gint bitrate;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
if (strcmp ("audio/x-adpcm", stname) == 0) {
if (!gst_structure_get_int (structure, "bitrate", &bitrate))
bitrate = 32000;
} else if (strcmp ("audio/G723", stname) == 0) {
bitrate = 24000;
} else if (strcmp ("audio/32KADPCM", stname) == 0) {
bitrate = 32000;
} else
goto invalid_caps;
switch (bitrate) {
case 16000:
encoding_name = g_strdup ("G726-16");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
2);
break;
case 24000:
encoding_name = g_strdup ("G726-24");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
3);
break;
case 32000:
encoding_name = g_strdup ("G726-32");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
4);
break;
case 40000:
encoding_name = g_strdup ("G726-40");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
5);
break;
default:
goto invalid_bitrate;
}
gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
gst_basertppayload_set_outcaps (payload, NULL);
g_free (encoding_name);
return TRUE;
/* ERRORS */
invalid_caps:
{
GST_ERROR_OBJECT (payload, "unknown caps specified");
return FALSE;
}
invalid_bitrate:
{
GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", bitrate);
return FALSE;
}
}
gboolean
gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg726pay",
GST_RANK_NONE, GST_TYPE_RTP_G726_PAY);
}

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/* GStreamer
* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
* Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTP_G726_PAY_H__
#define __GST_RTP_G726_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS typedef struct _GstRtpG726Pay GstRtpG726Pay;
typedef struct _GstRtpG726PayClass GstRtpG726PayClass;
#define GST_TYPE_RTP_G726_PAY \
(gst_rtp_g726_pay_get_type())
#define GST_RTP_G726_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G726_PAY,GstRtpG726Pay))
#define GST_RTP_G726_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G726_PAY,GstRtpG726PayClass))
#define GST_IS_RTP_G726_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G726_PAY))
#define GST_IS_RTP_G726_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G726_PAY))
struct _GstRtpG726Pay
{
GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpG726PayClass
{
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_g726_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_G726_PAY_H__ */